////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author        : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
//  SoundTouch audio processing library
//  Copyright (c) Olli Parviainen
//
//  This library is free software; you can redistribute it and/or
//  modify it under the terms of the GNU Lesser General Public
//  License as published by the Free Software Foundation; either
//  version 2.1 of the License, or (at your option) any later version.
//
//  This library is distributed in the hope that it will be useful,
//  but WITHOUT ANY WARRANTY; without even the implied warranty of
//  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
//  Lesser General Public License for more details.
//
//  You should have received a copy of the GNU Lesser General Public
//  License along with this library; if not, write to the Free Software
//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
//
////////////////////////////////////////////////////////////////////////////////

#ifndef AAFilter_H
#define AAFilter_H

#include "STTypes.h"
#include "FIFOSampleBuffer.h"

namespace soundtouch
{

class AAFilter
{
protected:
    class FIRFilter *pFIR;

    /// Low-pass filter cut-off frequency, negative = invalid
    double cutoffFreq;

    /// num of filter taps
    uint length;

    /// Calculate the FIR coefficients realizing the given cutoff-frequency
    void calculateCoeffs();
public:
    AAFilter(uint length);

    ~AAFilter();

    /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
    /// frequency (nyquist frequency = 0.5). The filter will cut off the
    /// frequencies than that.
    void setCutoffFreq(double newCutoffFreq);

    /// Sets number of FIR filter taps, i.e. ~filter complexity
    void setLength(uint newLength);

    uint getLength() const;

    /// Applies the filter to the given sequence of samples.
    /// Note : The amount of outputted samples is by value of 'filter length'
    /// smaller than the amount of input samples.
    uint evaluate(SAMPLETYPE *dest,
                  const SAMPLETYPE *src,
                  uint numSamples,
                  uint numChannels) const;

    /// Applies the filter to the given src & dest pipes, so that processed amount of
    /// samples get removed from src, and produced amount added to dest
    /// Note : The amount of outputted samples is by value of 'filter length'
    /// smaller than the amount of input samples.
    uint evaluate(FIFOSampleBuffer &dest,
                  FIFOSampleBuffer &src) const;

};

}

#endif
